September 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Strange DID routing behavior
by acheraime . 02 Sep '15

02 Sep '15

Hi, I'm experiencing a strange issue with freeswitch 1.4.21 running on CentOS 6.5. I have a trunk that terminates a few DIDs for 150 sip users. I'm using mod_python to dynamically generate all my dialplans. Everything have been working fine for three weeks (When I made this deployment) since Saturday, freeswitch start dropping calls. 95% of inbound calls are dropped with a "No route" message or an "invalid application" error. What's seems weird is the extension is generated. Below is a snippet of a log 2015-09-01 21:35:07.967396 [INFO] mod_dialplan_xml.c:635 Processing WIRELESS CALLER <603XXXXXXX>->813XXXXX in context public 2015-09-01 21:35:07.968396 [NOTICE] mod_python.c:212 Invoking py module: fscore.handler 2015-09-01 21:35:07.970394 [DEBUG] mod_python.c:283 Call python script 2015-09-01 21:35:07.970394 [INFO] switch_cpp.cpp:1328 This a dialplan request 2015-09-01 21:35:07.975403 [INFO] switch_cpp.cpp:1328 Call from outside network 2015-09-01 21:35:07.976400 [NOTICE] switch_cpp.cpp:1328 <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="dialplan" description="Dotcs automatically generated dialplan"> <context name="public"> <extension name="338_dotcs_route"> <condition field="destination_number" expression="^(813XXXXXXX)$"> <action application="log" data="notice INBOUND CALL FOR DID ${destination_number}"/> <action application="export" data="call_direction=inbound"/> <action application="transfer" data="1049 XML default"/> </condition> </extension> </context> </section> </document> 2015-09-01 21:35:07.976400 [DEBUG] mod_python.c:286 Finished calling python script Dialplan: sofia/external/603XXXXXXX@10.50.12.227:5065 parsing [public->338_dotcs_route] continue=false *Dialplan: sofia/external/603XXXXXXX@10.50.12.227:5065 <http://603XXXXXXX@10.50.12.227:5065> Regex (PASS) ] destination_number(813XXXXXXX) =~ /^(813XXXXXXX)$/ break=on-false* 2015-09-01 21:35:07.976400 [INFO] switch_core_state_machine.c:241 No Route, Aborting What do you guys think might be wrong? Thanks -- *Adolphe CHER-AIME*

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Python based Event Socket Library doesn't connect to remote FreeSWITCH
by Carlos Fererro 02 Sep '15

02 Sep '15

To all Event Socket Library FreeSWITCH gurus! Python implementation of the Event Socket Library doesn't seem to work with FreeSWITCH running on remote host. Only "localhost" or "127.0.0.1" has to be specified as the first parameter to the ESLconnection constructor to let client to successfully connect to the FreeSWITCH. Event if FreeSWITCH is running on the SAME machine (not even remote host) as the client and the "host" is specified with the IP address of the machine in the ESLconnection - no valid connection is getting established. I am pretty sure that such a fundamental thing can't not to work and I am just missing something simple. Any help is greatly appreciated. Thanks Carlos

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Connecting Freeswitch and Asterisk with outgoing registration
by Markus Bönke 02 Sep '15

02 Sep '15

Hello, I’ve connected freeswitch with an asterisk server via sip trunk with the following configuration in my test environment: Freeswitch side: <gateway name="sip.testprovider.com"> <param name="username" value="MyCustomSipTrunk1"/> <param name="password" value="easy123"/> <param name="extension" value="trunk"/> <param name="register" value="true"/> <param name="from_domain" value="sip.testprovider.com"/> </gateway> Asterisk side: sip.conf [MyCustomSipTrunk1] type=peer callerid="MyCustomSipTrunk1" <MyCustomSipTrunk1> host=dynamic nat=no username=MyCustomSipTrunk1 fromdomain=sip.testprovider.com directmedia=no disallow=all allow=gsm allow=ulaw allow=alaw secret=easy123 context=kamailio extensions.conf [kamailio] exten => _X.,1,Dial(SIP/${EXTEN}@MyCustomSipTrunk1,60,tr) If I send a call from asterisk to freeswitch, I can see the following in the log: 2015-09-02 00:06:28.877729 [DEBUG] sofia.c:9001 IP 192.168.1.202 Rejected by acl "domains". Falling back to Digest auth. 2015-09-02 00:06:28.877729 [WARNING] sofia_reg.c:2827 Can't find user [MyCustomSipTrunk1(a)sip.testprovider.com] from 192.168.1.202 You must define a domain called 'sip.testprovider.com' in your directory and add a user with the id="MyCustomSipTrunk1" attribute and you must configure your device to use the proper domain in it's authentication credentials. If I create the user in the directory for the domain, it works - but why do I need to create this user however the gateway is already already registered and authenticated with the asterisk server? Thanks Markus

2 2

SIP headers storage in core or sofia database.
by Pavel 02 Sep '15

02 Sep '15

Hello ! I've set up an ODBC connection for core and sofia profile like this: <param name="odbc-dsn" value="pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password='' options='-c client_min_messages=NOTICE' application_name='freeswitch'" /><param name="core-db-dsn" value="pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password='' options='-c client_min_messages=NOTICE' application_name='freeswitch'" /> Using PgAdmin I see some tables was created in Postgres DB. I want to retrieve all SIP invite headers from this DB. What table should I look for this information if, of course, it is stored at all ? Thanks. Regards,Pavel.

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FreeSWITCH Week in Review (Master Branch) August 22nd-29th
by Ken Rice 01 Sep '15

01 Sep '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1NLym47 FreeSWITCH Week in Review (Master Branch) August 22nd-29th Hello, again. This past week in the FreeSWITCH master branch we had 116 commits! Wow, the team was really busy this week! Our features for this week are: added getenv FSAPI to mod_commands, the verto communicator saw many improvements, and the beginnings of another new module! Mod_redis is being deprecated in favor of mod_hiredis! Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-8075 [mod_hiredis] Add mod_hiredis including support for redis limits and added support for raw redis commands. Added deprecation notices to mod_redis FS-8049 [mod_commands] Add getenv FSAPI FS-8036 [verto.js] Add chatCallback to $.verto.conf Improvements in build system, cross platform support, and packaging: FS-8061 [verto_communicator] Adding license to package.json FS-8047 [build] Fixed build error in mod_basic, mod_rtmp, mod_oreka, and mod_sangoma_codec due to using __FUNCTION__ on newer compilers FS-8054 [mod_rayo] Fixed a warning when building on Debian FS-8055 [build] Add confdir variable to freeswitch.pc FS-7966 [windows] Working msvc 2015 build support. Does not yet include video features. FS-8019 [debian] Excluded few modules that fail to compile from debian/bootstrap.sh, fixed the handling of -T and -t and added debian/apt_sources.list with necessary dependencies to build master, and updated debian/README.source FS-8058 [mod_vpx] Build correctly against libvpx that is not installed into default system library locations The following bugs were squashed: FS-8027 [verto communicator] Added watchTask flag to browserSync and add proper regex for replacements FS-8026 [verto_communicator] Added an auto-focus directive to both dial-pad and login so that enter will just work. On dial-pad useful to provide keyboard only input without the need to using the mouse FS-7995 [verto_communicator] Upon call recovery, emit an event on $rootScope so that controllers are able to properly clear states. FS-7945 [verto communicator] Use angular-prompt to ask the user to enter a text for the banner. If cancelled, nothing is done. FS-8045 [verto communicator] Make the folder structure compliant with AngularJS best practices and adjust build system. FS-7957 [verto_communicator] Make console less chatty by commenting liveArray updates and get initial state of the conference on liveArray boot event. FS-7979 [verto_communicator] Prompt for extension before transferring a conference member FS-8001 [verto_communicator] For this to work, passing in the parameter was missing FS-7979 [verto_communicator] Removed extra console.log and commented line FS-8025 [verto_communicator] Restored the blue background on the video controls and making icons white again, looking better. FS-8062 [verto_communicator] Fixed video controls tool-tips, now they are visible FS-8048 [verto_communicator] Fixed infinite reconnect after changing hostname and websocket url FS-8066 [verto communicator] Added encoded avatar url to userVariables so that mod_conference can use it when no video, or video mute FS-8018 [verto_communicator] Separation of concerns. Get storage service to manage all settings instead of vertoService FS-8043 [verto_communicator] Removed unnecessary calls to localStorage FS-8040 [verto_communicator] Check if we have a valid resolution reported before calling camera routines and hide controls if none are found FS-8092 [verto_communicator] If there is no data in localStorage, select best resolution for selected camera FS-7840 [verto_communicator] Use chatChannel to send and receive messages from conferences FS-8088 [verto_communicator] Call conference destroy upon hangup and on event destroy to properly unsubscribe from events FS-8046 [verto] Fixed for library not passing device preferences to dialog properly FS-8053 [verto] Don’t receive video on screen share FS-8059 [verto] Fixed typo when transferring party from conference FS-8060 [verto] Conditionally set video tag src to null for FF and empty string for others FS-8087 [verto] Fixed issue in camera selection on recovery, refactor to use localStorage, change res detection, reload on network change, pass resCheck callback into deviceParams, always make one basic call to getusermedia to ensure perms are ok, pass valid res to callback, make $.FSRTC.validRes available globally, sync minified, fix depth issue in cached json, test for valid cache before setting vars FS-8028 [mod_shout] Fixed random sockets being closed regression from FS-7621 FS-8029 [jitterbuffer] Fixed robotic sound when using jitterbuffer when buffer timestamps get behind that of the packet timestamps, such as when the source clock is out of sync with our clock FS-8056 [mod_voicemail] Fixed a segfault on vm_inject, regression from FS-7968 FS-7968 [mod_voicemail] Fixed verbose events FS-7942 [udptl] Fixed rare segfault on t.38 fax FS-8014 is a duplicate of this issue FS-8031 [dtls] Fixed delayed DTLS media due to changing ICE candidates FS-7903 [proxy_media] Fix Codec PROXY Exists but not at the desired implementation. 0hz 0ms 1ch error when using proxy media. FS-7989 [fixbug.pl] Add –author option FS-8037 [mod_sofia] Fixed so zrtp-passthru doesn’t activate unless the zrtp-hash is in the SDP FS-7135 [mod_sofia] Fixed response to re-invite with duplicate sdp (such as we get from session refresh) when soa is disabled to include an sdp. Fixed t.38 fax failure on session refresh FS-8050 [mod_av] Fixed a crash when streaming rtmp of desktop share FS-7640 [rtp] Fixed some comparisons in switch_rtp.c to be wraparound proof FS-8057 [core] Fixed a segfault when doing video call when built against libyuv but not libvpx FS-8069 [stun] Fixed ipv6 support missing in stun code FS-8071 [rtp] remove unnecessary auto adjust remote address rules when in ICE mode FS-8077 [mod_conference] Fix memory leak in record FS-8091 [core] Added some missing message names to session message name list (would have caused missing information in some log messages) FS-8093 [mod_silk] Remove giant stack allocation in switch_silk_decode And, this past week in the FreeSWITCH 1.4 branch we had 2 new commits merged in from master. And the FreeSWITCH 1.4.21 release is here! Go check it out! The following bugs were fixed: FS-7912 [mod_conference] Fixed for v1.4 branch to not allocate for conference cdr if conference cdr is disabled FS-8091 [core] Added some missing message names to session message name list (would have caused missing information in some log messages)

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Re: [Freeswitch-users] Broken silence with webrtc
by Gary Foreman 01 Sep '15

01 Sep '15

Ok so on a freshly built Debian install with FS 1.7 I get the exact same behaviour. I managed to stop the broken silence when using the originate command by adding absolute_codec_string=PCMA. My thoughts were it could be a transcoding thing but inbound calls from the Verto client (which have no issue) result in opus > g722 so cant be. The intermittent audio on verto calls is still a problem though. The most common issue is one way audio from the verto client. The recording has no audio from the verto side either so I guess the RTP stream isn't being sent. Does anyone have any pointers to where I can look when I reproduce the fault? It takes a while for the issue to occur but once I have the call active I need to find where in the chain it is falling down. No errors are reported in the Chrome or FS console. Thanks On Tue, Sep 1, 2015 at 4:36 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Broken silence with webrtc (Anthony Minessale) > > > ---------- Forwarded message ---------- > From: Anthony Minessale <anthony.minessale(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 1 Sep 2015 10:35:30 -0500 > Subject: Re: [Freeswitch-users] Broken silence with webrtc > Your findings contradict each other so much, I recommend you start over > from scratch. > Backup your configs. update to the latest master version of FS. If you > are on 1.4, nothing new will be done to mitigate webrtc issues. Set up a > box with the default configurations and retest. > > you may also want to try putting <X-PRE-PROCESS cmd="set" data= > "suppress_cng=true"/> in vars.xml > > > > On Tue, Sep 1, 2015 at 10:25 AM, Ken Rice <krice(a)freeswitch.org> wrote: > >> Just enable verto debugging in verto.conf.xml in your configs… it’ll >> print it right to the screen >> >> >> >> >> >> *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: >> freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Gary >> Foreman >> *Sent:* Tuesday, September 1, 2015 10:13 AM >> *To:* freeswitch-users(a)lists.freeswitch.org >> >> *Subject:* Re: [Freeswitch-users] Broken silence with webrtc >> >> >> >> Ok so the issue has been superseded by intermittent one-way / no audio. >> I'm getting it very intermittently (1 in every 30 calls or so) but I'm >> struggling to debug it as the traffic is encrypted and wireshark doesn't >> see it as rtp stream. >> >> >> >> Where is the best place to start debugging verto? I was previously using >> sip.js without any audio issues so it seems to be verto specific. >> >> >> >> On Tue, Sep 1, 2015 at 12:04 PM, < >> freeswitch-users-request(a)lists.freeswitch.org> wrote: >> >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users(a)lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request(a)lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner(a)lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> Today's Topics: >> >> 1. Re: Broken silence with webrtc (Stanislav Sinyagin) >> >> >> ---------- Forwarded message ---------- >> From: Stanislav Sinyagin <ssinyagin(a)gmail.com> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Tue, 1 Sep 2015 13:03:25 +0200 >> Subject: Re: [Freeswitch-users] Broken silence with webrtc >> >> is it running on a virtual machine? >> >> I found a strange effect that I could only reproduce in a VM, and never >> on physical hardware: >> https://freeswitch.org/jira/browse/FS-7805 >> >> under certain load, an originate command triggers a continuous distortion >> in another, running and unrelated, channel. >> >> It seems to be triggered by insufficient CPU resource at the moment of >> the origination. >> >> >> >> >> >> >> >> On Tue, Sep 1, 2015 at 12:49 PM, Gary Foreman <gaz.foreman(a)gmail.com> >> wrote: >> >> I've found that it occurs after any bridge, its not specific to the >> originate command. >> >> >> >> Would you require a wireshark trace or the output of the freeswitch >> console? >> >> >> >> The scenario below reproduces the issue ... >> >> >> >> Test extension >> >> >> >> <extension name="Inbound_Routing"> >> >> <condition field="destination_number" expression="^2003$"> >> >> <action application="export" data="dialed_extension=$1"/> >> >> <action application="export" data="transfer_ringback=$${uk-ring}"/> >> >> <action application="set" data="RECORD_TITLE=Title goes here"/> >> >> <action application="set" >> data="rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/> >> >> <action application="export" data="send_silence_when_idle=false"/> >> >> <action application="export" data="bridge_generate_comfort_noise=false"/> >> >> <action application="export" data="rtp_enable_vad_in=false"/> >> >> <action application="export" data="rtp_enable_vad_out=false"/> >> >> <action application="export" data="fire_talk_events=true"/> >> >> <action application="export" data="fire_not_talk_events=true"/> >> >> <action application="export" data="recording_id=${uuid}"/> >> >> <action application="set" data="media_bug_answer_req=true"/> >> >> <action application="answer" /> >> >> <action application="playback" >> data="tone_stream://L=100;%(400,200,400,450);%(400,2000,400,450)"/> >> >> </condition> >> >> </extension> >> >> >> >> I originate a call from a polycom handset using g722 to the extension >> above. >> >> I originate a call using the verto client to the extension above. >> >> >> >> I get the uuids of the channels using show channels and use uuid_bridge >> [uuid1] [uuid2] to merge the channels. >> >> >> >> On Tue, Sep 1, 2015 at 11:30 AM, < >> freeswitch-users-request(a)lists.freeswitch.org> wrote: >> >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users(a)lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request(a)lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner(a)lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> Today's Topics: >> >> 1. How send in To header anonymous and in P-Asserted-Identity >> caller-id-number (Alex Polischuk) >> 2. Loop play while wait DTMF digit (Dmitry Mordovin) >> 3. Broken silence with webrtc (Gary Foreman) >> 4. Re: Broken silence with webrtc (Brian West) >> 5. SIP profile not loading (Chris Young) >> 6. Playing with conditions (Dmitry Mordovin) >> 7. Re: Playing with conditions (Avi Marcus) >> >> >> ---------- Forwarded message ---------- >> From: Alex Polischuk <alxpol(a)gmail.com> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Mon, 31 Aug 2015 17:02:25 +0300 >> Subject: [Freeswitch-users] How send in To header anonymous and in >> P-Asserted-Identity caller-id-number >> >> Hi all, >> >> >> >> How I can define different users and domains in To and P-Asserted-Identity >> headers? >> >> >> >> Thanks, >> >> Alex >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: Dmitry Mordovin <d.mordovin(a)dwide.com> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Mon, 31 Aug 2015 18:24:08 +0400 >> Subject: [Freeswitch-users] Loop play while wait DTMF digit >> >> Hello >> >> This example play conf-pin.wav and wait DTMF. >> >> >> <extension name="play_and_get_digits example"> >> >> <condition field="destination_number" expression="^(1888)$"> >> >> <action application="play_and_get_digits" data="2 5 3 7000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav foobar \d+"/> >> >> <action application="log" data="CRIT ${foobar}"/> >> >> </condition> >> >> </extension> >> >> >> Is it possible to play WAV file in infinity loop and wait user DTMF? >> >> And how can I check DTMF input after user entered DTMF? >> >> In dialpeer, like. >> If ${DTMF} = 1 then bridge to XXX >> If ${DTMF} = 2 then play file and finish session >> >> >> >> Thank you. >> Dmitry >> >> >> >> ---------- Forwarded message ---------- >> From: Gary Foreman <gaz.foreman(a)gmail.com> >> To: freeswitch-users(a)lists.freeswitch.org >> Cc: >> Date: Mon, 31 Aug 2015 20:36:59 +0100 >> Subject: [Freeswitch-users] Broken silence with webrtc >> >> Hi all, >> >> >> >> Hoping someone can point me in the right direction because after several >> hours I'm out of ideas. >> >> >> >> I'm having an issue where the 2nd leg of a call that is bridged to a >> verto endpoint has broken silence. When there is no sound on the line the >> audio goes completely silent (all background noise is dropped) but around >> every third of a second it repeatedly cuts back in for a fraction, then >> goes completely silent again. This only happens when the call is created >> using the originate command. >> >> >> >> Can anyone give me an idea of where to look next? I have a wireshark >> trace that during playback shows the audio cutting out periodically on the >> 2nd leg during periods of silence. The 1st leg is using webrtc encryption >> and I cant decode the stream. >> >> >> >> Thanks in advance! >> >> >> >> >> >> >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: Brian West <brian(a)freeswitch.org> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Mon, 31 Aug 2015 14:40:17 -0500 >> Subject: Re: [Freeswitch-users] Broken silence with webrtc >> >> Please provide logs and samples of how you originate this, sounds like >> Voice Activity Detection possibly. >> >> >> >> On Mon, Aug 31, 2015 at 2:36 PM, Gary Foreman <gaz.foreman(a)gmail.com> >> wrote: >> >> Hi all, >> >> >> >> Hoping someone can point me in the right direction because after several >> hours I'm out of ideas. >> >> >> >> I'm having an issue where the 2nd leg of a call that is bridged to a >> verto endpoint has broken silence. When there is no sound on the line the >> audio goes completely silent (all background noise is dropped) but around >> every third of a second it repeatedly cuts back in for a fraction, then >> goes completely silent again. This only happens when the call is created >> using the originate command. >> >> >> >> Can anyone give me an idea of where to look next? I have a wireshark >> trace that during playback shows the audio cutting out periodically on the >> 2nd leg during periods of silence. The 1st leg is using webrtc encryption >> and I cant decode the stream. >> >> >> >> Thanks in advance! >> >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> >> -- >> >> *Brian West* >> brian(a)freeswitch.org >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: >> /r/freeswitch <https://www.reddit.com/r/freeswitch> >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> >> >> ---------- Forwarded message ---------- >> From: Chris Young <Chris.Young(a)enghouse.com> >> To: "freeswitch-users(a)lists.freeswitch.org" < >> freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Tue, 1 Sep 2015 07:31:00 +0000 >> Subject: [Freeswitch-users] SIP profile not loading >> >> Hi all, >> >> >> >> Recently, we've begun experiencing a strange problem whereby the first >> SIP profile to be loaded gets 'stuck' and never actually completes its >> initialisation. This always seems to affect the first profile only, so if I >> have profiles named (for example): >> >> >> >> dummy.xml >> >> external.xml >> >> internal.xml >> >> >> >> then 'dummy' would fail to load but 'external' and 'internal' would be >> fine. No error messages are output to the logs but preliminary >> investigation suggests that the profile thread is becoming blocked for some >> reason. The specified IP address is valid and available and there are no >> other processes using the requested port. FreeSWITCH comes up successfully >> but 'sofia status' shows only 'external' and 'internal'. At this point, I >> can use 'sofia profile dummy start' and the profile loads correctly so it >> appears to be valid. >> >> >> >> Has anybody else seen this kind of behaviour or know what could be >> causing it? >> >> >> >> Many thanks, >> >> Chris >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: Dmitry Mordovin <d.mordovin(a)dwide.com> >> To: "freeswitch-users(a)lists.freeswitch.org" < >> freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Tue, 01 Sep 2015 12:56:08 +0400 >> Subject: [Freeswitch-users] Playing with conditions >> Hello >> >> <section name="dialplan" description="Local call"> >> <context name="public"> >> <extension name="local_1"> >> >> <condition field="destination_number" expression="^(\d{4})$"> >> <action application="answer"/> >> >> <action application="play_and_get_digits" data="1 1 3 7000 # >> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav >> foobar \d+"/> >> <action application="log" data="CRIT ${foobar}"/> >> >> <condition field="${foobar}" expression="1$"> >> <action application="log" data="CRIT >> --------------1------------------"/> >> <anti-action application="log" data="CRIT --------------not >> 1------------------"/> >> </condition> >> >> <action application="log" data="CRIT ---- last ${foobar}"/> >> </condition> >> >> </extension> >> </context> >> </section> >> >> Why inner condition not works? >> >> >> >> >> ---------- Forwarded message ---------- >> From: Avi Marcus <avi(a)avimarcus.net> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Tue, 1 Sep 2015 10:30:00 +0000 >> Subject: Re: [Freeswitch-users] Playing with conditions >> >> Short answer: Each extension only has 1 set of conditions. >> >> The condition evaluating foobar is run *before* it gets set. >> >> After the play_and_get_digits you should transfer/execute_extension to a >> new extension that will evaluate ${foobar} >> >> >> -Avi Marcus >> BestFone >> >> >> >> On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin <d.mordovin(a)dwide.com> >> wrote: >> >> Hello >> >> <section name="dialplan" description="Local call"> >> <context name="public"> >> <extension name="local_1"> >> >> <condition field="destination_number" expression="^(\d{4})$"> >> <action application="answer"/> >> >> <action application="play_and_get_digits" data="1 1 3 7000 # >> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav >> /invalid.wav foobar \d+"/> >> <action application="log" data="CRIT ${foobar}"/> >> >> <condition field="${foobar}" expression="1$"> >> <action application="log" data="CRIT >> --------------1------------------"/> >> <anti-action application="log" data="CRIT --------------not >> 1------------------"/> >> </condition> >> >> <action application="log" data="CRIT ---- last ${foobar}"/> >> </condition> >> >> </extension> >> </context> >> </section> >> >> Why inner condition not works? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > <http://freeswitch.org/g+>* > > ClueCon Weekly Development Call > ☎ sip:888@conference.freeswitch.org ☎ +19193869900 > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

2 1

Delay and exec after answer
by Dmitry Mordovin 01 Sep '15

01 Sep '15

Hello Is it possible to execute some api with delay after answer received? Trying api_on_answer_1='sleep 5000',api_on_answer_2='log -------delayed 5sec after answer------' but 'api_on_answer_2' executed at once Any ideas?

3 4

CHANNEL_HANGUP_COMPLETE not firing‏
by Gaurav Khanna 01 Sep '15

01 Sep '15

Hi, I am using ESL Inbound connection to get CDR information. I have added "ALL" as filter so that I get all the events which are raised by freeswitch. But for some reason "Channel_hangup_complete" event is not raised. "Channel_Hangup" and "Channel_Execute_complete" works correctly. Also tried "Plain event Channel_hangup_complete" command but that also doesn't work. Also tried linger() function but that also doesn't work. Is there any settings in config folder or in var file which I need to set. Also tried following settings in dialplan Thanks, Gaurav

4 7

Freeswitch 1.6 video recording issue using uuid_record
by Ankit Sharma 01 Sep '15

01 Sep '15

We are using freeswitch as media server in integration with kamailio. So the purpose of using freeswitch is to record the two-way video calls between the legs so without any recording freeswich application or API being used my video and audio is working just perfect (running on default media handling mode i.e. proxy_media & bypass media not used), late-negotiation is disabled, zrtp-passthru is set to false and disable-transcoding is set to true after exploring all over web over my recording concern i got to know about the following APP/APIs: record, record_session, record_fsv, record_mp4, uuid_record (however i want to record in .mp4 format) so following the very first step I looked for "video-media-bug" branch and got to know it is merged in freeSWITCH 1.6 master, so used the 1.6 master , and now the vide-media-bug branch is contained within my current repo now using all above mentioned applications i'am not getting my recording done, here i'am sharing what's happening with uuid_record being used as my kbridge dialplan is: <extension name="kbridge"> <condition field="destination_number" expression="^kb-(.+)$"> <action application="set" data="proxy_media=false"/> <action application="set" data="call_timeout=30"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="sip_invite_domain=192.168.5.99"/> <action application="set" data="voicemail_authorized=${sip_authorized}"/> <action application="export" data="sip_contact_user=ufs"/> <action application="export" data="api_args=${uuid_setvar ${uuid} enable_file_write_buffering=false"/> <action application="export" data="api_result=${uuid_record ${uuid} start /usr/local/freeswitch/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_$1_${caller_id_number}_${destination_number}.mp4}"/> <action application="bridge" data="sofia/$${domain}/$1(a)192.168.5.99"/> <action application="answer"/> <action application="voicemail" data="check auth $${voicemail_profile} default $${domain} $1"/> </condition> </extension> and the fs_cli log says: 2015-08-21 13:19:50.980274 [NOTICE] switch_channel.c:1089 New Channel sofia/internal/102(a)192.168.5.99 [c1d25c8d-4e6e-458c-abe0-c40b89bc3d68] 2015-08-21 13:19:50.990233 [INFO] mod_dialplan_xml.c:637 Processing 102 <102>->kb-101 in context public 2015-08-21 13:19:50.990233 [NOTICE] switch_ivr.c:1877 Transfer sofia/internal/102(a)192.168.5.99 to XML[kb-101@default] 2015-08-21 13:19:50.990233 [INFO] mod_dialplan_xml.c:637 Processing 102 <102>->kb-101 in context default 2015-08-21 13:19:51.000223 [INFO] switch_ivr_async.c:2339 Sending early media 2015-08-21 13:19:51.000223 [INFO] switch_core_media.c:6543 Activating VIDEO RTCP PORT 5244 interval 2000 mux 1 2015-08-21 13:19:51.000223 [NOTICE] mod_sofia.c:2296 Pre-Answer sofia/internal/102(a)192.168.5.99! 2015-08-21 13:19:51.010223 [INFO] mod_native_file.c:101 Opening File [/usr/local/freeswitch/recordings/2015-08-21-13-19-51-in.G722] 16000hz 2015-08-21 13:19:51.010223 [INFO] mod_native_file.c:101 Opening File [/usr/local/freeswitch/recordings/2015-08-21-13-19-51-out.G722] 16000hz 2015-08-21 13:19:51.010223 [NOTICE] switch_channel.c:1089 New Channel sofia/internal/101(a)192.168.5.99 [5127cf3a-c7ec-4038-a033-f287b2999a79] 2015-08-21 13:19:51.220289 [NOTICE] sofia.c:6817 Ring-Ready sofia/internal/101(a)192.168.5.99! 2015-08-21 13:19:55.240230 [INFO] switch_core_media.c:6543 Activating VIDEO RTCP PORT 0 interval 2000 mux 0 2015-08-21 13:19:55.240230 [NOTICE] sofia.c:7576 Channel [sofia/internal/101(a)192.168.5.99] has been answered 2015-08-21 13:19:55.240230 [NOTICE] switch_ivr_originate.c:3526 Channel [sofia/internal/102(a)192.168.5.99] has been answered 2015-08-21 13:20:04.180275 [NOTICE] sofia.c:952 Hangup sofia/internal/102(a)192.168.5.99 [CS_EXECUTE] [NORMAL_CLEARING] 2015-08-21 13:20:04.290279 [NOTICE] switch_ivr_bridge.c:874 Hangup sofia/internal/101(a)192.168.5.99 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2015-08-21 13:20:04.290279 [NOTICE] switch_core_session.c:1657 Session 29 (sofia/internal/101(a)192.168.5.99) Ended 2015-08-21 13:20:04.290279 [NOTICE] switch_core_session.c:1661 Close Channel sofia/internal/101(a)192.168.5.99 [CS_DESTROY] 2015-08-21 13:20:04.300232 [NOTICE] switch_core_session.c:1657 Session 28 (sofia/internal/102(a)192.168.5.99) Ended 2015-08-21 13:20:04.300232 [NOTICE] switch_core_session.c:1661 Close Channel sofia/internal/102(a)192.168.5.99 [CS_DESTROY] now in the recordings directory i got 2 files(can be seen in the log ) the files hav the extension .G722, which is not supposed to be the required output since uuid_record is an API I tried using uuid_record in console so started the call (after commenting the record api in dialplan) and it gave me this error: -ERR Cannot locate session! so help me out over this concern thanks in advance :) Ankit

2 1

call disconnect in 60 sec
by Bharathkumar J 01 Sep '15

01 Sep '15

Hi, we are ususing FS 1.7 with sipml5 ,when make outgoing video calls through sipml5 it disconnect exactly 60 sec *Regards,* *J.Bharathkumar*

2 1